similar to: Marketing 101

Hi, I am working on developing a professional Marketing Materials for my systems. I plan on using a very good(expensive) company to do that so splitting the costs with several people would be nice. Let me know if you are interested on taking part in it. robert -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm kind of at a loss to diagnose problems like this, yet we get them a lot. - The ATA (Thomson 784 in this particular case) is logged into the Asterisk server. 'sip show peer' shows their IP address, port, and useragent. - The ATA is connected directly to the internet (no NAT, but the sip configuration has nat=always) and logs in to our server, which is also directly connected to the

hello friends, I want to install Asterisk on a Debian machine. I need to download the sources or just with apt-get install is enought??? thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070320/227a3b32/attachment.htm

I have an interesting task for my son's lacrosse team... it is the time-old telephone tree... I am pretty sure someone has already done this w/*, why re-invent the wheel?... a) coach calls in leaves a msg, others call in retrieve the msg b) coach calls in leaves a msg, kicks of a call to every parent plays msg c) coach calls in leaves a msg, kicks off a call to every parent, checks for

hi, I'm getting registration errors I can't debug... [Mar 23 11:07:20] NOTICE[2952]: chan_iax2.c:7344 socket_process: Registration of 'host2' rejected: 'Registration Refused' from: '10.10.10.82' I was getting a 'Cause Code: 29' INV,POKE,...,REJ but I can't duplicate that level of debugging again in the CLI> on host15 10.10.10.15

I use qualify in sip.conf and need to setup a trigger when asterisk sees it as unreachable, so that I can either drop a call file, or send an email, or both. How can I do that? Thank you

hi, I have 2 asterisks with the following configuration. asterisk server 1 (S1) has an user 9002 asterisk server 2 (S2) has an user 9003 Both users can make call to each other without problem. Now I add both users to both servers, i.e. asterisk server 1 (S1) has users 9002,9003 asterisk server 2 (S2) has users 9002,9003 When 9002 dials 9003, Dial(SIP/9003@S2) or visa versa. Both processes

Hi i'm student and my final project is related to Voip. I have Asterisk almost fully configured. The next step is to accept login of users, that data is in Universitys database which uses ActiveDirectory and also Ldap and Kerberos. It's possible? I don't want authentications in sip.conf, but in other remote database. The problem is i don't have ideas how to start with. I would

hello there all, if i have a script that writes drop files into /var/spool/asterisk/outgoing asterisk picks up the file and initiates the call just fine. however, what is supposed to happen if more than one gets dropped in there within like a second. Will it wait till the first is complete to initiate the second ? Do they dissapear ? thanks shawn -------------- next part -------------- An HTML

Hi all, I upgrade my asterisk from 1.2.11 to 1.4.2 changing my entire dial plan but I have the following errors and I'm not able to call anymore. Do you know what can I have to do? My Asterisk is connected to a patton with a SIP trunk. [Mar 22 10:19:03] WARNING[16462]: chan_sip.c:12311 handle_response: Remote host can't match request BYE to call

I recently upgraded from asterisk 1.2.13 to 1.4.2 and am looking at using the users.conf file to setup my users, before i was using real time SIP which worked fine. However when i create a user in users.conf i am unable to register the user form a softphone, however that same softphone can still register a different the users i currently have setup form the sip.conf from real time. i've

I have set up a script that ensures certain services are up on my Asterisk box (Trixbox 2.0). I would like it to trigger a wake-up call if certain conditions aren't meant. How might I accomplish this from the shell? -- Donovan Niesen Customer Contact Services www.yourccsteam.com

Hi! As the upstream of my DSL-connection is very slow, I'd like my sip-phones to use iLBC to connect to my *. My gateway provider only allows ulaw. Hence, I'd like to use the follwing setup: SIP-phone <--iLBC--> Asterisk <---ulaw----> PSTN-Gateway I get the following error: "Unable to find a codec translation path from ilbc to ulaw" Setup SIP-phone: disallow=all

Hello, 1. Is it possible to install several SIP softphones on the same PC, have them registered to the same Asterisk server and attribute to each softphone a specific extension, ringtones or call forwarding rules ? 2. Is possible to do the same with SIP hardphones ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am looking at ways to harden my asterisk install to prevent computer related issues from happening. I am concerned about about disk write cache. That seems to be a major source of hard drive corruption on power failure. Hard Drive corruption is simply unacceptable for the 99.999% uptime requirements of my Asterisk install that needs to be as reliable as a proprietary PBX. Of course I will be

Hello Fellow Asterisk Mailing ListMembers, When I tried to download the latest version of Asterisk this is what I get: http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.0.tar.gz Opening fileinfo database failed http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.0.tar.gz Opening fileinfo database failed Where are all the latest Asterisk 1.4.x source files? Thanks in advance, -E

Anyone have an idea to re master centos,in other worlds I have an asterisk on centos with all libraries and modules,how can I make it as an iso image ? Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of

This is an email to all New York based Asterisk users. For some time it's been bugging me that we don't have a local contact point/user community. If you are involved in Asterisk and in NY/NJ shoot me an email, I'm going to try and revitalize either meetup.com or some other shared environment for Asterisk users in NY. Shoot me an email and once I get an idea of how many

Hi, I know that Asterisk doesn't support Instant Messaging, but I'm trying to use the AGI function RECEIVE TEXT to implement a kind of IM service. I have a sip softphone that tries to send a message to an active channel and the AGI script that expect to receive the text through the STDIN. Two problems arise: First: How can I say to asterisk to get the message? (I see on CLI console that

I was just curious to what your redundancy solution is. I have considered many options, so I thought I would share and get an idea for what others are doing. My setup is two different locations with a 10MB WLAN fiber link between the two. Each location has it's own PRI as well. I have considered and tested many options this last year or so. 1) Using hearbeat and drbd to monitor the